Hello, I have a working FS instance I am using as a redirect server (to serve up LNP requests). It works fine on calls with invites that have SDP and does not work with invites without SDP. I enabled 3pcc to true thinking that would fix the issue. Version info is FreeSWITCH Version 1.0.6 (hacked-20100921T052029Z). With the console log level set to debug the only thing I see is this message (just before returning a 480): freeswitch@lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT] switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout waiting for next instruction in CS_NEW! I also do not ever see FS to an xml http req (which is how we control the dialplan). freeswitch@lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at 14:57:50.934431: ------------------------------------------------------------------------ INVITE sip:19033226103@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 Max-Forwards: 70 Contact: <sip:X.X.X.X:5060> To: <sip:19033226103@X.X.X.X:5060> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE Date: Tue, 09 Nov 2010 15:29:25 GMT Supported: 100rel User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152 Content-Length: 0 ------------------------------------------------------------------------ send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060> Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Content-Length: 0 ------------------------------------------------------------------------ send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060>;tag=t7ycB64mX33mQ Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE" Content-Length: 0 Remote-Party-ID: "19033226103" <sip:19033226103@X.X.X.X
;party=calling;privacy=off;screen=no
------------------------------------------------------------------------ Here is the sofia profile: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="sip-ip" value="X.X.X.X"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <param name="inbound-bypass-media" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <param name="disable-transcoding" value="true"/> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Any thoughts / ideas / help would be greatly appreciated. Thanks! SDR
The default behaviour for INVITEs without SDP is to reject the call because media cannot be negotiated. It's possible to accept these INVITEs by setting the following parameter on the SIP profile: <param name="enable-3pcc" value="true"/> See these previous discussions from the archive: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg03504.h... http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg11825.h... Warm regards, -Steve On 10 November 2010 15:22, SIC FS LIST <sicfslist@gmail.com> wrote:
Hello, I have a working FS instance I am using as a redirect server (to serve up LNP requests). It works fine on calls with invites that have SDP and does not work with invites without SDP. I enabled 3pcc to true thinking that would fix the issue. Version info is FreeSWITCH Version 1.0.6 (hacked-20100921T052029Z). With the console log level set to debug the only thing I see is this message (just before returning a 480): freeswitch@lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT] switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout waiting for next instruction in CS_NEW! I also do not ever see FS to an xml http req (which is how we control the dialplan). freeswitch@lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at 14:57:50.934431: ------------------------------------------------------------------------ INVITE sip:19033226103@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 Max-Forwards: 70 Contact: <sip:X.X.X.X:5060> To: <sip:19033226103@X.X.X.X:5060> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE Date: Tue, 09 Nov 2010 15:29:25 GMT Supported: 100rel User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152 Content-Length: 0 ------------------------------------------------------------------------ send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060> Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Content-Length: 0 ------------------------------------------------------------------------ send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060>;tag=t7ycB64mX33mQ Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE" Content-Length: 0 Remote-Party-ID: "19033226103" <sip:19033226103@X.X.X.X>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ Here is the sofia profile: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="sip-ip" value="X.X.X.X"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <param name="inbound-bypass-media" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <param name="disable-transcoding" value="true"/> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Any thoughts / ideas / help would be greatly appreciated. Thanks! SDR _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
And I really should read emails properly rather than jumping to conclusions... you've already tried that. :o) Try posting full debug-level logs for the entire length of the call. Context is important, not just the CRIT error but what happened leading up to it. That'll also probably show the HTTP request problem. What're you using - mod_xml_curl? My hunch is it's something about codec negotiation since you're setting bypass media, late negotiation and disabling transcoding. Try removing the disable-transcoding parameter. The effect of that is to only offer to the B-leg the codec from the A-Leg. Since the A-Leg has no codec selected because there's no SDP to offer one and you're delaying selecting a codec (late negotiation) then there's perhaps no codec to offer to the B-leg. -Steve On 10 November 2010 15:41, Steven Ayre <steveayre@gmail.com> wrote:
The default behaviour for INVITEs without SDP is to reject the call because media cannot be negotiated.
It's possible to accept these INVITEs by setting the following parameter on the SIP profile:
<param name="enable-3pcc" value="true"/>
See these previous discussions from the archive: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg03504.h... http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg11825.h...
Warm regards, -Steve
On 10 November 2010 15:22, SIC FS LIST <sicfslist@gmail.com> wrote:
Hello, I have a working FS instance I am using as a redirect server (to serve up LNP requests). It works fine on calls with invites that have SDP and does not work with invites without SDP. I enabled 3pcc to true thinking that would fix the issue. Version info is FreeSWITCH Version 1.0.6 (hacked-20100921T052029Z). With the console log level set to debug the only thing I see is this message (just before returning a 480): freeswitch@lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT] switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout waiting for next instruction in CS_NEW! I also do not ever see FS to an xml http req (which is how we control the dialplan). freeswitch@lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at 14:57:50.934431: ------------------------------------------------------------------------ INVITE sip:19033226103@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 Max-Forwards: 70 Contact: <sip:X.X.X.X:5060> To: <sip:19033226103@X.X.X.X:5060> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE Date: Tue, 09 Nov 2010 15:29:25 GMT Supported: 100rel User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152 Content-Length: 0 ------------------------------------------------------------------------ send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060> Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Content-Length: 0 ------------------------------------------------------------------------ send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060>;tag=t7ycB64mX33mQ Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE" Content-Length: 0 Remote-Party-ID: "19033226103" <sip:19033226103@X.X.X.X>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ Here is the sofia profile: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="sip-ip" value="X.X.X.X"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <param name="inbound-bypass-media" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <param name="disable-transcoding" value="true"/> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Any thoughts / ideas / help would be greatly appreciated. Thanks! SDR _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Steve, Thanks for the assistance. The entire console log after setting console loglevel 9 is as follows: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Everything else is just the SIP messaging. It's like FS never gets the call from sofia for some reason. I tried the following (your comments around the media make sense for sure): -- remove the disable transcoding (this did not work) -- remove the other codec parameters (this did not work) To clarify what the box does: -- takes an invite -- responds with a 302 redirect with a modified contact header that looks like $dn;npdi=yes;rn=$rn (if there is an rn)@$host info. This is done with using mod_xml_curl to provide the dynamic dialplan. There are not ever any b legs. Every call is an invite, 100 trying, 302, ACK. In this case we get an invite, a 100 trying and that's it. Doing an ngrep -d any -qW byline port 80 shows that there is not a xml_curl req and on the console at loglevel 9 the only message from FS is this: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Sofia spits out all of the SIP messaging but that's it. It looks nothing like a normal call when I send the SDP. It's a little odd. Here is the modified profile after commenting out all of the info: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001.sipinterchange.com"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="4.71.122.205"/> <param name="sip-ip" value="4.71.122.205"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <!-- <param name="inbound-bypass-media" value="true"/> --> <!-- <param name="inbound-late-negotiation" value="true"/> --> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <!-- <param name="disable-transcoding" value="true"/> --> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Thanks again for the help. SDR
"Everything else is just the SIP messaging." And dialplan/call processing... that's precisely the context it's useful to see. -Steve On 10 November 2010 16:22, SIC FS LIST <sicfslist@gmail.com> wrote:
Steve, Thanks for the assistance. The entire console log after setting console loglevel 9 is as follows: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Everything else is just the SIP messaging. It's like FS never gets the call from sofia for some reason. I tried the following (your comments around the media make sense for sure): -- remove the disable transcoding (this did not work) -- remove the other codec parameters (this did not work) To clarify what the box does: -- takes an invite -- responds with a 302 redirect with a modified contact header that looks like $dn;npdi=yes;rn=$rn (if there is an rn)@$host info. This is done with using mod_xml_curl to provide the dynamic dialplan. There are not ever any b legs. Every call is an invite, 100 trying, 302, ACK. In this case we get an invite, a 100 trying and that's it. Doing an ngrep -d any -qW byline port 80 shows that there is not a xml_curl req and on the console at loglevel 9 the only message from FS is this: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Sofia spits out all of the SIP messaging but that's it. It looks nothing like a normal call when I send the SDP. It's a little odd. Here is the modified profile after commenting out all of the info: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001.sipinterchange.com"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="4.71.122.205"/> <param name="sip-ip" value="4.71.122.205"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <!-- <param name="inbound-bypass-media" value="true"/> --> <!-- <param name="inbound-late-negotiation" value="true"/> --> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <!-- <param name="disable-transcoding" value="true"/> --> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Thanks again for the help.
SDR _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
It's like FS never gets the call from sofia for some reason. I tried the following (your comments around the media make sense for sure): -- remove the disable transcoding (this did not work) -- remove the other codec parameters (this did not work) If you think sofia's not passing the call to FS you can check the internal processing of the sofia stack using "sofia loglevel all 9". That'll show it picking up the call and any errors within the stack before it reaches FS. -Steve On 10 November 2010 16:22, SIC FS LIST <sicfslist@gmail.com> wrote:
Steve, Thanks for the assistance. The entire console log after setting console loglevel 9 is as follows: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Everything else is just the SIP messaging. It's like FS never gets the call from sofia for some reason. I tried the following (your comments around the media make sense for sure): -- remove the disable transcoding (this did not work) -- remove the other codec parameters (this did not work) To clarify what the box does: -- takes an invite -- responds with a 302 redirect with a modified contact header that looks like $dn;npdi=yes;rn=$rn (if there is an rn)@$host info. This is done with using mod_xml_curl to provide the dynamic dialplan. There are not ever any b legs. Every call is an invite, 100 trying, 302, ACK. In this case we get an invite, a 100 trying and that's it. Doing an ngrep -d any -qW byline port 80 shows that there is not a xml_curl req and on the console at loglevel 9 the only message from FS is this: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Sofia spits out all of the SIP messaging but that's it. It looks nothing like a normal call when I send the SDP. It's a little odd. Here is the modified profile after commenting out all of the info: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001.sipinterchange.com"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="4.71.122.205"/> <param name="sip-ip" value="4.71.122.205"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <!-- <param name="inbound-bypass-media" value="true"/> --> <!-- <param name="inbound-late-negotiation" value="true"/> --> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <!-- <param name="disable-transcoding" value="true"/> --> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Thanks again for the help.
SDR _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks for the help. I apparently can't read and made this harder on myself than needed. Once I changed the enable-3pcc param to proxy it works fine now. Thanks again. Sorry for wasting your time. SDR
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SIC FS LIST
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Steven Ayre